In Asterisk while try to call in between sip phones Aborted

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In Asterisk while try to call in between sip phones Aborted

Postby surya » Wed Jul 04, 2012 5:10 am

HI,
I need your help
Actually i have installed Asterisk in my gateway having OpenWrt. My gateway supports OpenWrt. I have configured in sip.conf and extension.conf file for two users .Then i run asterisk by command
# asterisk -cvvvvvvv
My both the sip phones registered successfully. But when i dialed the 2nd phone by dialing 301 ( i have configured)
the asterisk aborted and i am not able to call .For your reference last few lines of logs captured

== Registered application 'Playback'
app_playback.so => (Sound File Playback Application)
Asterisk Ready.
*CLI> == Using SIP RTP CoS mark 5
-- Executing [301@imonitor:1] Dial("SIP/user2-00000000", "sip/user1,20,rt") in new stack
[Jul 4 10:54:23] WARNING[1450]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'sip' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/user2-00000000' status is 'CHANUNAVAIL'
Aborted
@OpenWrt:/etc/asterisk#

please help me how i will make call via asterisk
surya
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Re: In Asterisk while try to call in between sip phones Aborted

Postby leemason » Wed Jul 04, 2012 7:23 am

It's not enough just to register both SIP phones in sip.conf. You also need to build a dial plan for each number in extensions.conf. This will tell Asterisk what to do when you dial a specific number.
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Re: In Asterisk while try to call in between sip phones Aborted

Postby david55 » Wed Jul 04, 2012 11:35 am

The trace shows he has got a dialplan. It is complaining about sip.conf, or about outgoing registration or qualify problems.

An abort is almost always the result of the malloc package discovering memory corruption and is always a bug unless mixed versions of packages were somehow installed. Abort is what you get when the assert library routine fails an assertion.

The version of Asterisk isn't stated, so I can't tell whether bug reports will still be accepted for it, although he would be advised to get the very latest released version on the 1.8 or 10 series (in case the bug has already been fixed), and make sure that the binaries are unoptimised and unstripped (which are pre-conditions for submitting any crash bug report).

Sometimes, you can identify the primary fault leading to a corruption by reports about locking problems (you probably need thread debugging enabled for that), but otherwise the standard protocol for memory corruption bug reports involves running it interpretively under VALGRIND. NB this is very slow, and might not be supported on non-Intel platforms.

If you are not very familiar with Asterisk, I would advise getting the system working on CentOS on an i386 platform, before working with router hardware.
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Re: In Asterisk while try to call in between sip phones Aborted

Postby david55 » Wed Jul 04, 2012 11:44 am

Also, AsteriskNow is an architecture dependent system image that will only run on Intel Architecture, so either this is the wrong forum, or you need to contact the creator of the OpenWRT clone.
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Re: In Asterisk while try to call in between sip phones Aborted

Postby leemason » Thu Jul 05, 2012 2:22 am

Please post the sip.conf content for the two extensions plus the dial plan and we can have a look.
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